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This chapter contains the following information about the Cisco SIP IP phone:
Session Initiation Protocol (SIP) is the Internet Engineering Task Force's (IETF's) standard for multimedia conferencing over IP. SIP is an ASCII-based, application-layer control protocol (defined in RFC 2543) that can be used to establish, maintain, and terminate calls between two or more end points.
Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call.
SIP provides the capabilities to:
Conferences can consist of two or more users and can be established using multicast or multiple unicast sessions.
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Note The term conference means an established session (or call) between two or more end points. In this document, the terms conference and call are used interchangeably. |
SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user agent can function in one of the following roles:
Typically, a SIP end point is capable of functioning as both a UAC and a UAS, but functions only as one or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the UA that initiated the request.
From an architecture standpoint, the physical components of a SIP network can also be grouped into two categories: clients and servers. Figure 1-1 illustrates the architecture of a SIP network.
Cisco SIP IP phones 7960s (hereafter referred to as Cisco SIP IP phones) are full-featured telephones that can be plugged directly into an IP network and used very much like a standard private branch exchange (PBX) telephone. The Cisco SIP IP phone is an IP telephony instrument that can be used in VoIP networks.
The Cisco SIP IP phone model terminals can attach to the existing in place data network infrastructure, via 10BaseT/100BaseT interfaces on an Ethernet switch. When used with a voice-capable Ethernet switch (one that understands Type of Service [ToS] bits and can prioritize VoIP traffic), the phones eliminate the need for a traditional proprietary telephone set and key system/PBX.
The Cisco SIP IP phone complies with RFC 2543.
Figure 1-2 illustrates physical features of the Cisco SIP IP phone:
In addition to the physical features illustrated in Figure 1-2, the Cisco SIP IP phone also provides the following:
The "Route" attribute of the Template tag in the dial plan template file can be used to indicate which proxy (default, emergency, FQDN) that the call should be initially routed to. For example, to configure an emergency proxy, specify value of the "Route" attribute as "emergency".
When the primary proxy does not respond to the INVITE message sent by the Cisco SIP IP Phone after the configured number of retries, the Cisco SIP IP Phone sends the INVITE to the backup proxy. This is independent from which proxy is defined in the "Route" attribute in the dial plan template used.
The Cisco SIP IP Phone does not have to register with the backup proxy. All interactions, such as authentication challenges, with the backup proxy is treated the same as the interactions with the primary proxy .
The backup proxy is only used with new INVITE messages. Once the backup proxy is used, it is active for the duration of the call.
The location of the backup SIP proxy can be defined as an IP address in the default configuration file. See proxy_backup and proxy_backup_port parameters in Modifying the Default SIP Configuration File in "Managing Cisco SIP IP Phones" .
An optional emergency SIP proxy can be configured with the "Route" attribute of the Template tag in the Dial Plan template file. See "Support of user-defined proxy routing".
When an emergency SIP proxy is configured and a call is initiated, the phone generates an INVITE message to the address specified in the proxy_emergency parameter. The emergency proxy is used for the call duration.
The location of the emergency proxy can be defined as an IP address in the default configuration file. See proxy_emergency and proxy_emergency_port parameters in Modifying the Default SIP Configuration File in "Managing Cisco SIP IP Phones" .
DNS SRV is the Domain Name Server RR used to locate servers for a given service.
SIP on Cisco's SIP IP Phones use DNS SRV query to determine the IP address of the SIP Proxy or the Redirect Server. The query string generated is in compliance with RFC2782, and prepends the protocol label with an underscore "_"; as in "_protocol._transport.". The addition of the underscore reduces the risk of the same name being used for unrelated purposes.
Also in compliance with RFC 2782 and the draft-ietf-sip-srv-01 spec. is that the system can remember multiple IP addresses and use them properly. In the draft-ietf-sip-srv-01 spec, it is assumed that all proxies returned for the SRV record are equivalent such that the phone can register with any of the proxies and initiate a call using any other proxy.
VAD can be enabled or disabled with enable_vad parameter. Value 0 for disable, and value 1 for enable. See enable_vad parameter in Modifying the Default SIP Configuration File in "Managing Cisco SIP IP Phones" .
Three-way conferencing supports one phone conferencing with two other phones by providing mixing on the initiating phone. To set up a 3-way conference call, see documentation on Making Conference Calls in "Getting Started with the Cisco IP Phone 7960". See Release Note for limitations.
If the INVITE message contains an Alert-Info header, distinctive ringing is invoked, Format of the header is "Alert-info: x". "x" can be any number. This header is only received by the phone and is not generated by the phone.
Dinstinctive ringing is supported when the phone is idle or during a call. In the idle mode, the phone rings with a different cadence. The selected ringing type plays twice with a short pause in between. In call-waiting mode, two short beeps are generated instead of one long beep.
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Note For information on how to use the standard telephony features and URL dialing, refer to the Getting Started Cisco IP Phone 7960 and Quick Reference Cisco IP Phone 7960 documents. |
The Cisco SIP IP phone supports the following standard protocols:
DNS is used in the Internet for translating names of network nodes into addresses. SIP uses DNS to resolve the host names of end points to IP addresses.
DHCP is used to dynamically allocate and assign IP addresses. DHCP allows you to move network devices from one subnet to another without administrative attention. If using DHCP, you can connect Cisco SIP IP phones to the network and become operational without having to manually assign an IP address and additional network parameters.
The Cisco SIP IP phone complies with the DHCP specifications documented in RFC 2131. By default, Cisco SIP IP phones are DHCP-enabled.
ICMP is a network layer Internet protocol that enables hosts to send error or control messages to other hosts. ICMP also provides other information relevant to IP packet processing.
The Cisco SIP supports ICMP as it is documented in RFC 792.
IP is a network layer protocol that sends datagram packets between nodes on the Internet. IP also provides features for addressing, type-of-service (ToS) specification, fragmentation and reassembly, and security.
The Cisco SIP IP phone supports IP as it is defined in RFC 791.
RTP transports real-time data (such as voice data) over data networks. RTP also has the ability to obtain Quality of Service (QoS) information.
The Cisco SIP IP phone supports RTP as a media channel.
SDP is an ASCII-based protocol that describes multimedia sessions and their related scheduling information.
The Cisco SIP IP phone uses SDP for session description.
SNTP sychronizes computer clocks on an IP network. The Cisco SIP IP phones use SNTP for their date and time support.
TFTP allows files to be transferred from one computer to another over a network.
The Cisco SIP IP phone uses TFTP to download configuration files and software updates.
UDP is a simple protocol that exchanges data packets without acknowledgments or guaranteed delivery. SIP can use UDP as the underlying transport protocol. If UDP is used, retransmissions are used to ensure reliability.
The Cisco SIP IP phone supports UDP as it is defined in RFC 768 for SIP signaling.
For the Cisco SIP IP phone to successfully operate as a SIP endpoint in your network, your network must meet the following requirements:
For more information about configuring IP, refer to Cisco IOS IP and IP Routing Configuration Guide.
For more information about configuring VoIP, refer to the Cisco IOS Release 12.1 Multiservice Applications Configuration Guide for the appropriate access platform. For more information about configuring SIP VoIP, refer to the Enhancements to SIP for VoIP on Cisco Access Platforms.
The Cisco SIP IP phone has connections for connecting to the data network, for providing power to the phone, and for connecting a headset to the phone. Figure 1-3 illustrates the connections on the Cisco SIP IP phone.
The Cisco SIP IP phone has two RJ-45 ports that each support 10/100 Mbps half- or full-duplex Ethernet connections to external devices—network port (labeled 10/100 SW) and access port (labeled 10/100 PC). You can use either Category 3 or 5 cabling for 10 Mpbs connections, but use Category 5 for 100 Mbps connections. On both the network port and access port, use full-duplex mode to avoid collisions.
Use the network port to connect the phone to the network. You must use a straight-through cable on this port. The phone can also obtain inline power from the Cisco Catalyst switch over this connection. See the "Connecting to Power" section for details.
Use the access port to connect a network device, such as a computer, to the phone. You must use a straight-through cable on this port.
The Cisco SIP IP phone can be powered by the following sources:
This module sends power on pins 1 & 2 and 3 & 6.
This module sends power on pins 4, 5, 7, and 8.
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Note Only the network port (labeled 10/100 SW) supports inline power from the Cisco Catalyst switches. |
For redundancy, you can use the Cisco AC adapter even if you are using inline power from the Cisco Catalyst switches. The Cisco SIP IP phone can share the power load being used from the inline power and external power source. If either the inline power or the external power goes down, the phone can switch entirely to the other power source.
To use this redundancy feature you must set the inline power mode to auto on the Cisco Catalyst switch. Next, connect the un-powered Cisco SIP IP phone to the network. After the phone powers up, connect the external power supply to the phone.
The Cisco SIP IP phone supports a four or six-wire headset jack. Specifically, the Cisco SIP IP phone supports the following Plantronics headset models:
The Volume and Mute controls will also adjust volume to the earpiece and mute the speech path of the headset. The headset activation key is located on the front of the Cisco SIP IP phone.
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Note When using a headset, an amplifier is not required. However, a coil cord is required to connect the headset to the headset port on the back of your Cisco IP Phone 7960. For information on ordering compatible headsets and coil cords for the Cisco IP phone 7960, see http://cisco.getheadsets.com or http://vxicorp.com/cisco. |
To function in the IP telephony network, the Cisco SIP IP phone must be connected to a networking device, such as a Catalyst switch, to obtain network connectivity.
The Cisco SIP IP phone has an internal Ethernet switch, which enables it to switch traffic coming from the phone, access port, and the network port.
If a computer is connected to the access port, packets traveling to and from the computer and to and from the phone share the same physical link to the switch and the same port on the switch.
This configuration has these implications for the VLAN configuration on the network:
You can resolve these issues by isolating the voice traffic onto a separate VLAN on each of the ports connected to a phone. The switch port configured for connecting a phone would have separate VLANs configured for carrying:
Isolating the phones on a separate, auxiliary VLAN increases the quality of the voice traffic and allows a large number of phones to be added to an existing network where there are not enough IP addresses.
For more information, refer to the documentation included with the Cisco Catalyst switch.
Posted: Sat Nov 29 15:21:23 PST 2003
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